Vodia PBX

Put a star character into the replacement in the dial plan where you want to reject the further processing of the dial plan.

Well in the old times we used that 150 KB template for Polycom which somewhere must have included the settings that told the phone to use so-and-so many lines. I guess we need to dig that settings out again, and then insert it into the new template--if the settings is set. I think the default behavior now is good and we should change it only if the user has specifically added the settings to the extension.

For the version 4 releases, it is easier to just do the workaround. It is very difficult for us to make version 4 builds as the build environment is not available to us any more. Those servers are not accessible to us any more.

The old "video" worked like this: The phone receives a request to display a XML page. In that page was a link to an image on the PBX. The phone would fetch then an image from the PBX. Then there was another link in the XML that says, fetch another page. Then the phone would load a new XML, which contains a new link for a new image. Then the phone would download that file. And everything gets challenged for security reasons. And so on... "moving pictures" by generating at least two HTTP requests to the PBX per picture frame! It is easy to see that this was a dirty hack to get something working at all. The 870 now supports MJPEG, which is also a simple standard, but which does not put any load on the PBX. The phone can then fetch the images from the camera directly. This is definitively the way to go. You have to put the URL into the phone's settings, either directly on the phone or change the template for the phone on the PBX.

Ohhhhh.... Okay now I understand you. I thought you wanted difference extensions on the same phone. So what you want are several private lines on the phone. You actually have that by default, even though there is only one button associated with it. Did you try setting the "lines" parameters in the registrations tab? At least in the old pbxnsip days, that provisioned several lines on the phone; though with the latest version that could conflict with the explicit button assignments (but worth a quick try). But again, I would personally not put several lines on the precious buttons on the phone, instead use the phone user interface to toggle between the different calls on the phone.

Well you need to remove and add the MAC to all extensions that share the same MAC. But apart from that, you did it right. The generated folder is not read, only written and should help you figure out what was sent to the device. You can double check the macs directory if the PBX got it right.

The problem is related to the "last one who saves the MAC wins" problem that we have already discovered and even fixed. It is already fixed and available in the 5.0.10i version (http://snomone.com/d...ion-5.0.10i.xml). After upgrading, remove the MAC and save them again.

Okay, finally tried this on my own. Seems like you are right, there is something wrong with the "Not Allowed" entry! Try putting in a replacement, for example *. We'll fix this in the next release.

You can download the latest exec also with wget http://snomone.com/d...rwin10-5.0.10i.

Did you try to run it as root? Without it, you might not have the permissions to open sockets. Also, check what other applications are using the ports using netstat -a. Maybe there is even another PBX already running, ps auxww|grep pbxctrl will show you that.

We are about to release 5.1.0; but in the meantime you can take http://snomone.com/downloads/snomONE/version-5.0.10i.xml. The socket binding was a problem if the port was used before, either by the PBX itself or another service. We have smartened up the logic to grab the necessary ports. The process must run with root permissions, if you want to try to run it manually, use sudo.

As for PSTN gateways, top of my head appear Patton, AudioCodes, Grandstream, Vegastream (now Sangoma), Quintum, Mediatrix, and also Digium has now stand-alone PSTN gateways. Also you should take a look at what SIP trunk providers offer if you want to use your Internet access to terminate VoIP calls.

Oh. One more thing. If you are using country code 1, then all numbers are fed in a normalized form into the dial plan. Domestic calls are always presented in 10 digit format (no leading 1), all other number are presented with 011.... That means that [2-9][0-9][0-9]976xxxx should work for you.

We tried it out a couple of years ago. At the end of the day, the question is if a all-in-one box is the way to go. We tried the approach of the snom ONE plus; however the results was for us that it is easier to keep the PSTN gateway and the server topic separated. Not to mention the CS410, which we also tried a couple of years ago. Honestly, out appetite for a box with PSTN termination included is low after those experiences. There is a market of PSTN gateways with great quality and sometimes also great prices. Today, you get 8 x FXO, E1, T1, J1, PRI and so on without any problems. For us, it is better to focus on having a great server hardware and having a great PSTN gateway hardware and not mix it up.

If you are using complex pattern, you must try to match also the domain part. For example, 6173998147@domain.com. Your pattern does not match the @ character. This this: ^1[2-9][0-9][0-9]976.* (note the dot) Or try this "simple" pattern: 1[2-9][0-9][0-9]976xxxx

For the mini, the number of G.729A calls is limited to 5. Because this is an embedded platform with a relatively small processor, you can't extend that too much anyway. I would say 10 calls is definitively the limit.

You should see that in the status screen. Adding features to a license was something that we introduced in version 5.

The thing with G.729A is that is must be licensed per channel. If you are running out of channels, you get that message unless you allow other codecs. That is a feature...

That's a general problem. The phone can generate a missed call entry only if it got a incoming call; however you can actually miss call without the phone ringing. That's why these two lists are not in sync. On 3xx phones that were provisioned with PnP, we have put the missed call list on a button (the snom button), so that the phone always loads the missed call list from the PBX. Then the lists should always be in sync.

In system admin mode, edit the domain settings (icon on the right in the domain list)

You can check at https://www.snomone.com/versions if there is anything that you'll need. For example, fax2email did not exist in version 4 so you won't have a problem there. Missing ACD or the limitation of two calls could be something you'll miss; however buying additional calls is not expensive if you need additional ten calls.

We have made builds for Windows 32/64, Linux Debian 32/64, CentOS 32/64, MacOS (Intel and PPC) and ARM (snom ONE mini). For those who would like to move to the beta, you can use the link http://snomone.com/downloads/snomONE/version-5.0.10i.xml to automatically upgrade your system. As usual, please make a backup before the upgrade, so that if there should be anything that you don't like, you can always move back. Also, as we have made a lot of changes to the webpages, please rename the html directory and the webpages directory, so that you can get a fresh start with the changed pages. This would be especially important for the provisioning. Release Notes will follow.

Unter Email/Email Einstellungen (System Administrator-Mode), http//pbx/reg_email.htm.

We are currently at 10j... Public available release hopefully available soon...

Check out e.g. snom_3xx_phone.xml and snom_760_phone.xml. There is a article about this topic on the snom wiki http://wiki.snom.com/Category:HowTo:Background_Image; it shows the problem: it is not enough to just have one image; you need to have the image available in different resolutions and formats. You can put the images into the tftp folder of the PBX. As URL you can then use http://ip-of-the-pbx/tftp/image-name.jpg. This way, you don't have to use an external HTTP server to serve the images.